Hi,
I have asterisk 1.2.12 running on my server. Everything seems to be
working fine on it. It has an IAX connection to the
terminator/orignator. Again, everything seems to be fine.. calls
come in and go out. However, it seems that after a call has been up
for several minutes audio will go o
I don't believe there is any quick & simple way of doing this.
You would need to add the column in the DB and modify cdr.c.
I'm sure someone out there has a "step by step" doc on how to do this. You may try the #asterisk channel on irc.
bp
On 10/19/06, unplug <[EMAIL PROTECTED]> wrote:
Thanks
I have installed the libtiff(3.5.7),spandsp-0.0.2pre24,app_txfax and
app_rxfax,asterisk-1.2.12.1 on the CentOS 4.2.
I set sip.conf like this:
[sip_local]
host=192.168.2.111
type=friend
dtmfmode=rfc2833
canreinvite=no
insecure=very
Thanks!!
Just one more question. Can I do the same "add fieldname=1" if I add
a field "fieldname" in the cdr table to perform the same action?
On 10/19/06, William Piper <[EMAIL PROTECTED]> wrote:
In cdr_mysql.conf add "userfield=1" under the globals setting.
bp
On 10/18/06, unplug <[EMAIL P
In cdr_mysql.conf add "userfield=1" under the globals setting.
bp
On 10/18/06, unplug <[EMAIL PROTECTED]> wrote:
I want to set some custom data in the field of userfield in table CDRas following.exten => s,19,Set(CDR(userfield)=1234)
exten => s,20,Dial(SIP/1234)However, the userfield doesn't get
I want to set some custom data in the field of userfield in table CDR
as following.
exten => s,19,Set(CDR(userfield)=1234)
exten => s,20,Dial(SIP/1234)
However, the userfield doesn't get update after making the call.
After that, I relocate the command as following.
exten => s,19,Dial(SIP/1234)
e
In the astdb, there will be a record once a sip user register.
However, I found that the record will still stay in the astdb even
when the user not register for a long long time. Can I refresh the
astdb by some command such that it will get the update status of the
system?
Thanks.
___
Group
Any
known problems with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface
or the vmail.cgi script?
I'm
unable to see voicemails via the web even though the MWI is flashing and
if I look in /var/spool/asterisk/voicemail/default/100/INBOX
I
do see msg files in
Group
Any known problems
with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi
script?
I'm unable to see
voicemails via the web even though the MWI is flashing and if I look in
/var/spool/asterisk/voicemail/default/100/INBOX
I do see msg files
in that folder.
H
If you have four pstn telephone numbers (eg, 444-1212, 444-1213,
444-1214, and 444-1215) from your telco, then call the telco and have
them implement call forwarding on each of the four lines. You might also
verify they provide a "call forwarding on busy" function for those lines.
After they h
all thanks for the replies. i know what to do now. thanks.John Novack <[EMAIL PROTECTED]> wrote: What provider?Pots lines?SOME providers will provide hunting on residential lines, but not all, and most probably not 4 lines. Hunting does not require any thing more than the providers switch programm
okay, i undrestand what you guys are saying. thanks alot.Brent Franks <[EMAIL PROTECTED]> wrote: > i would think i would need one DID with multiple> simultaneous connections.Hello, you can't setup a DID per se on an analog line.Essentially what you want is 4 regular POTS line in a hunt group.The f
rich, thanks for replying. i assume your talking about enabling call forward and call forward on busy from my vsp side. i dont quite grasp everything else that your saying, can you explain in laymen terms. thanks.Rich Adamson <[EMAIL PROTECTED]> wrote: Christopher Corn wrote:> i plan on buying 4
Christopher Corn wrote:
i plan on buying 4 residential lines for our small office and i was
giving some thought. we'd like to have one main number that can transfer
calls to the other lines. but seeing that i have 4 different individual
lines with different numbers, im not seeing hows thats pos
i would think i would need one DID with multiple
simultaneous connections.
Hello, you can't setup a DID per se on an analog line.
Essentially what you want is 4 regular POTS line in a hunt group.
The first channel will be your incoming number. If busy, it will roll
to line 2, line 3, etc unti
Some telcos will set up your lines so that the calls will go to
available lines, rather than the first one.
PaulH
AsteriskIT
On Mon, 2006-09-11 at 16:01 -0700, Christopher Corn wrote:
> i plan on buying 4 residential lines for our small office and i was
> giving some thought. we'd like to have o
What provider?
Pots lines?
SOME providers will provide hunting on residential lines, but not all,
and most probably not 4 lines. Hunting does not require any thing more
than the providers switch programmed to do so, but most will not do more
than two lines.
VOIP it all depends again on the prov
i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats possible, without tying up a li
Yea, but even with that you can't run a macro from the CLI. You'd need the manager API to do this.
bp
On 8/30/06, Tim St. Pierre <[EMAIL PROTECTED]> wrote:
Sort of. There is a command line argument to the asterisk process that runsit's arguments as CLI commands. You could write a shell script
Hello, I need some advice on the following problem I'm trying to solve:
At the office we are using 7940s as our phones, connected to an asterisk
box via SIP. Pretty standard setup, nothing fancy. Everyone has an
extension that comes out as a single line button on the phones, with the
second line u
I don't believe that the CLI was ever designed to do what you are asking. You should check out the manager API to do this type of stuff.
bp
On 8/30/06, unplug <[EMAIL PROTECTED]> wrote:
Hi,In CLI, I can issue a dial command. How can I run a macro in CLI?Is it possibe?Thanks.
___
Hi,
In CLI, I can issue a dial command. How can I run a macro in CLI?
Is it possibe?
Thanks.
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ot;gc" [EMAIL PROTECTED]
Date: Mon, 28 Aug 2006 07:40:34 -0700
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about context for incoming calls
> I am new to asterisk. After studying the book and the tutorial on the
> Internet, I am still confued about how to use cont
gc wrote:
I am new to asterisk. After studying the book and the tutorial on the
Internet, I am still confued about how to use context for incoming calls.
Here is my question:
If I create s extensions in two different contexts for incoming calls
which one will be used? When a call comes in, whic
I am new to asterisk. After studying the book and
the tutorial on the Internet, I am still confued about how to use context for
incoming calls.
Here is my question:
If I create s extensions in two different contexts
for incoming calls which one will be used? When a call comes in, which conte
Hi,
I have set a queue 5000 with a agent logged in. When an user
dial 5000, will ring and then answer the call. Queue function
will execute and recording is started. However, the recorded wav file
is not a valid file with just a few bytes. Anyone can help me to
enable the recording
When I put a call from an H323 phone to an asterisk box equiped with oh323
I cannot hear any ring tone on the phone (NetMeeting).
When call is answered everything is OK, i can hear and the other person too
can hear me.
I found this:
https://skylab.inaccessnetworks.com/mantis/view.php?id=79
which
Hi,
When I run iax2 show netstats... what is each side? Obviously Local
is me and Remote is the other side... but which is which direction?
That being is local me --> remote or is local remote --> me?
LOCAL -
REMOTE ---
Do you mean the patch can use to replace asterisk DB by ARA?
On 7/24/06, Matt Riddell (NZ) <[EMAIL PROTECTED]> wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
> Unplug, I'm sure there are other people with better ideas but if you
> see on sineapps, I remember someone having writt
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
> Unplug, I'm sure there are other people with better ideas but if you
> see on sineapps, I remember someone having written a patch which
> seperates out the the sip registry into a new table. If this is stable
Save you searching:
http://ww
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written a patch which
seperates out the the sip registry into a new table. If this is stable
and tested, then you might want to use that with an ARA configuration
and have all your aster
Thanks! Actually, I want to share the asterisk DB using multiple
asterisks. So I use NFS to share the whole directory
/var/lib/asterisk in order to share files including astdb of asterisk.
However, there is not what I expected. Say, UA1 registers asterisk1
and UA2 registers asterisk2. Asterisk
On Sat, 2006-07-22 at 11:34 +0800, unplug wrote:
> That's right. We can find many information about Realtime (ARA). But
> I can't find any information talking about asterisk DB. Where does it
> store data? As I said before, there is a file
> /var/lib/asterisk/astdb. Does it used to store data
That's right. We can find many information about Realtime (ARA). But
I can't find any information talking about asterisk DB. Where does it
store data? As I said before, there is a file
/var/lib/asterisk/astdb. Does it used to store data in asterisk DB?
Anymore information about asterisk DB th
cemail+ODBC+storage
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of unplug
Sent: Friday, July 21, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question about asterisk DB
Hi,
What is the mechanism of aster
EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of unplug
Sent: Friday, July 21, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question about asterisk DB
Hi,
What is the mechanism of asterisk DB? In folder /var/lib/asterisk,
there is a file c
Hi,
What is the mechanism of asterisk DB? In folder /var/lib/asterisk,
there is a file called astdb. Does it used for storing data of
asterisk DB?
thanks,unplug
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Hi,
I am using realtime function to query the table.
Table sipprop
name,number,forward
Peter,1234,0
May,4321,1
RealTime(sipprop|name|Peter) and it can simple get the output of
number if the record exist.
If I issue RealTime(sipprop|name|John), asterisk will show "No
Realtime Matches Found". W
Kai Ober wrote:
Eric "ManxPower" Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try to r
Eric "ManxPower" Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try to remember, if i'm
Kai Ober wrote:
Eric "ManxPower" Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS*
Eric "ManxPower" Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
wondering
Kai
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On 6/11/06, James Harper <[EMAIL PROTECTED]> wrote:
[.snip.]
My dialplan in the pap2 is:
(<:0>S0)
Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what
Kai Ober wrote:
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
SIPura supports it, Cisco ATAs support it. I assume that Cisco phones
support it.
I don't know about Grandstream devices since they a
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
Regards
Kai
It's called "hotline" or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for "hotline" in phone docs
Zap: immedi
Hi,
I want to know about the content of ast db. It is like a registry
of the asterisk to store information about register users. The
similar user register information will be stored in DB in ARA. I want
to verify that when user sends a register request and it is valid,
asterisk will capture th
Hi All:
I got inconsistent Agent complete events, I placed two calls to dial
the same queue 3666,
agent complete event has right information and time stamp in first
call. Second call was terminated at 09:24:59,
the agent complete event came in at wrong time at 09:24:42.
Does anyone know why? Pleas
Hi,
Does anyone know what asterisk do during the register & invite
process using ARA (realtime) with Nat enabled?
Say there is an UA1 send an register request to asterisk. Asterisk
will parse the register request header (in which source file?). It
will get the necessary information & update t
ED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
SuttonSent: Monday, June 26, 2006 11:15 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Question
about ring groups and ext. busy in call
I have a ring group set up with 3
extensions… we’ll use 14, 15 and 16.
When a call comes in, it rings
I have a ring group set up with 3 extensions… we’ll
use 14, 15 and 16.
When a call comes in, it rings all three extensions. If one
particular extension already is on the phone, it completely skips that phone
and only rings the other 2. Example to explanation sake is:
Call comes in
Does the CALLERID(all) also set the ANI Information, or
does the Set CALLERID(ani) also have to be called?
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ht
All:
I tested echo test by dialing *43 under Asterisk configured by FreePbx
by using x-lite softphone. I could not figure out how the call is routed
to context from-internal. In sip_additional.conf, I have three
extensions defined as 2826, 2800 and 2801, which all are defined context
as from-inter
James Harper wrote:
> So... asterisk can't tell the difference between 's' for 'no extension
> dialled', and when 's' was actually the name of the extension dialled...
> is this the expected behaviour?
>
>
I surely hope so, you can refer to it as such in the extensions.conf as
well (with goto et
> On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
> > Ideally I would have liked the pap2 to have done the same as
'immediate'
> > when talking about fxo, capi, misdn, etc, but I couldn't get it to
> > automatically dial nothing. A '0' was the best I could do. If anyone
> > knows how to put
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
> Ideally I would have liked the pap2 to have done the same as 'immediate'
> when talking about fxo, capi, misdn, etc, but I couldn't get it to
> automatically dial nothing. A '0' was the best I could do. If anyone
> knows how to put it into im
> James Harper wrote:
> > Easy to do on the Linksys PAP2, if that helps. The functionality
> > probably depends on the make and model of the phone... maybe if you
gave
> > those details as well?
> >
> > James
> >
> Fantastic, this may solve the problem In the mail I've just posted
> (which hasnt' a
James Harper wrote:
> Easy to do on the Linksys PAP2, if that helps. The functionality
> probably depends on the make and model of the phone... maybe if you gave
> those details as well?
>
> James
>
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
I
>Easy to do on the Linksys PAP2, if that helps. The functionality
> probably
depends on the make and model of the phone... maybe if you gave
> those details
as well?
>
> James
>
Well, there's the rub. I don't have any of the
hardware yet. I am looking at the various options before buying anyth
It's called "hotline" or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for "hotline" in phone docs
Zap: immediate=yes, runs exten => when phone is picked up
Cisco and others: Look up PLAR
[EMAIL PROTECTED] wrote:
Basically, I am looking to set up an extension which
PROTECTED]
> Sent: Sunday, 11 June 2006 10:42
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Question setting up a "bat phone" extension.
>
> Basically, I am looking to set up an extension which will be used as a
> "help-line".
> I want
Basically, I am looking to set up an extension which will be used as a
"help-line".
I want it to function kind of like the bat phone from the old Batman series,
where Commissioner Gordon would pick up the extension in his office and it
would ring the phone back at Wayne's mansion. Is there a way
Hi there, SIP more commanly used and it is a openstandard. Meanwhile IAX2 is a protocol on asterisk. I dont think it will effect the cpu resources cause they are bid with the same codecs like G711 and etc..so if you used SIP or IAX2 it also refer to the same codecs...so dont think it will take a lo
Dear Friends and Supporters!
I have successfully installed the asterisk at home,
and it is working perfectly with sip and iax2 users. However, I am
wondering that should we use only sip or only iax2? If we use both
of them, does it take a lots of cpu resources to translate between sip
Your Dial command must have both T and t in it to be able to transfer
both incoming and outgoing calls.
in features.conf, you can change # for blind transfer to ## -- this lets
you use # in banks and voicemails and other auto attendants.
Moj
Matt wrote:
Hi I'm a little confused here...
Am Wednesday 26 April 2006 20:43 schrieb Wai Wu:
> If I download zaptel-1.2.5, do I still have to apply the
> zaptel-1.2.5-patch?
no.
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> Asterisk-Users mailing list
> To UNSUBSCRIBE or upd
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
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Hi,
If I am interfacing with a legacy PBX system a few questions.
#1 What do I need to do to configure 1 port on a dual port card as
pri_cpe and another as pri_net? Do I just change my config half-way
through the zaptel.conf file?
#2 When I setup span=1,1,0,esf,b8zs doesn't the esf indicate
Tielin Xu wrote:
I'd like to use FreePBX, it seems some setup inconsistency with
Asterisk RealTime, do you know any other good admin tool for Asterisk?
FreePBX is not designed to work with Asterisk RealTime. I don't know of
a GUI to configure RealTime myself. :)
--
National Manager - Specia
Avi:
I got login failed message from x-lite.
I checked asterisk log, I got following:
res_config_mysql.c MySQL RealTime: Retrieve SQL: SELECT * from
sip_peers where name = '2826'
res_config_mysql.c MySQL RealTime: Everything is fine.
res_config_mysql.c MySQL RealTime: Failed to query database. Ch
Tielin Xu wrote:
I noticed that there is no ip address stored for my softphone in Mysql,
how does the Asterisk know which computer my softphone is running? I
checked the config files, no softphone registrations in sip.conf.
freePBX stores your phone information in sip_additional.conf and does
Hi All:
I used FreePBX to configure Asterisk, and tables are create in MySQL by
using FreePBX install script.
I created two x-lite softphone accounts by using FreePBX, they are
stored in table sip as friend.
I followed wiki doc to edit the extconfig.conf file.
I can not get those two softphone to
Hi I'm a little confused here... trying to setup a parking lot...
lot is setup but how do I send calls to the parkinglot? If I
allow #700 transfer, it seems I can only transfer on inbound calls...
if I use a T in my dialplan I can only transfer on outbound calls...
additionally pressing #
t; From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: Tuesday, April 11, 2006 8:15 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Question on clicking
> >
> > This isn't an a
l Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Tuesday, April 11, 2006 8:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Question on clicking
>
> This isn't an asterisk question,
This isn't an asterisk question, more of a telco question. Hopefully
someone on this list involved with telco can answer it.
I have a client who is using asterisk with analog phone lines. There
is a bad clicking noise on the line. "Click Click Click".
We have narrowed it down to bei
ll
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, April 11, 2006 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question on clicking
This isn't an asterisk question, more of a telco qu
On Sonntag, 9. April 2006 9:58 Tele Cost Price Reducer wrote:
> hi Ronald,
>
> i would use a CallerIDNum authentication, based on the Asterisk
> Database to solve it.
>
> then you do not need any verification.
Dangerous. CallerIDs can be easily faked in some countries using VoIP
providers.
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk Database to solve it.
then you do not need any verification.
you just build a list of approved numbers in the database and then have a context checking the whitelist.
if you need more help, let me know,
Mickey
On 4/8/06
Lists,
Hi, good day, i was being task to create a DISA access for internal
purpose of the company, i'm having a problem to work with it with
authentication, but i think it's really a straight forward thing to do,
can someone enlight me on this. thanks
sample code snippet
exten => 5,Got
Hi, I have questions about the Polycom 601 and side card
1) In the side card the lights all time off... But all functions it's ok.
I need help with extension module of polycom... All works fine... But lights
not work So... I don't know when any person or extension is busy...
Any ideas?
Augenstine
Sent: March 17, 2006 8:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Question about meetme app
My mistake. Locking a conference from the CLI does prevent any additional
callers from connecting. But AFAIK locking the conference does not
I have two simple questions regarding compiling Zaptel based on the
confusing (to a kernel newbie) instructions at
http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation.
The instructions say:
For a Linux 2.6 kernel you may need to:
modify the EXTRAVERSION statement in Makefile so i
l Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
> Augenstine
> Sent: March 17, 2006 5:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Question about meetme app
>
> A locked conferen
Sent: March 17, 2006 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about meetme app
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
> I hav
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
> I have a quick question about the MeetMe app. A locked conference means
> what exactly?
>
> A) That people can't join anymore
> B) That everyone is muted
I have a quick question about the MeetMe app. A locked conference means
what exactly?
A) That people can't join anymore
B) That everyone is muted except the admin
Follow-up question
If the answer above is A, how do you accomplish B?
Mick
___
--Band
n Behalf Of Devon
Watkins
Sent: Thursday, March 16, 2006 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question about advanced IVR
Hi all,
I am a fairly new * user, so please bear with me. I have been scoring
the net for this info, but have had little
> 2)Read the book "Asterisk, The future of telephony".
> You can buy it or download it for free. I dont have
> the link to it but if some one else does please post
> it.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
___
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I Would reccomend learning asterisk bit by bit. It's
the only way to do it. Here are several tools that you
can use.
1)Get [EMAIL PROTECTED] Its a great resource for
starters. You can learn a lot from it. You can get it
at http://asteriskathome.lists.digium.com
2)Read the book "Asterisk, The future
Hi all,
I am a fairly new * user, so please bear with me. I have been scoring
the net for this info, but have had little success.
I am looking for any tips and info on how to build a "information
delivery" ivr app. By that I mean it will simply play prerecorded sound
files and respond to key
What exactly do you need? A digium card could be anything from one
pstn line, to multiple t1 lines, to who knows what else. And serial
number authentication...what's this for? Does a user dial in, enter
in a serial, then get access to something? Like a calling card, or
something completely diff
Hey all,
I have a client whose previous programmer ditched. I'm his webmaster,
and now he wants me to have an asterisk system set up for serial
number authentication...and I have a digium card from the previous
guy. Are there hosts that will set this up for me? ie, rack space
somwhere? Are the
Hello,
This question is probabely recurrent, i apologize, but i haven't found a
limpid explanation (for me) in mail list, google, and hum source
code):
When use the Command DIAL to ring a group, WHERE is stored the name of
the 'winner' who pick up the call ? ($variable = ?), and, step beyo
Hi,
We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where
users register with an OpenSER cluster (2 nodes currently).
When they request PSTN they are forwarded to * where they have entries
in SIP realtime database. This ensures that they get their correct
CallerID and contex
Hello,
I have a situation where I need to differentiate between registrations
by users where there might be clashes on the left hand side (username)
portion of the SIP From URI. (for a multi-domain virtual hosting system)
It seems that only the username portion is used for SIP authentication,
My prior box would do this too, with exactly 99.987793 almost every time
as you saw. It only had a wildcard, x100p. actually, clone maybe. The
other notable difference is that I see you're using apic. have you
tried with the noapic kernel option to see if that changes your results?
Anyway
Yes,
But I'm reading on the opposite side, or always I loose 1 sample?
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest -v
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample i
On Mon, Jan 16, 2006 at 09:00:28PM -0500, Carlos Alperin wrote:
> Another request make me test my t1 card, which has no quality problems, but
> all that I get is:
>
>
>
> [EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
>
> Opened pseudo zap interface, measuring accuracy...
>
> 99.987793% 99.987793%
Carlos Alperin wrote:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
Is anything wrong about this?
I never get 100.00 % most of the times.
No problem there. Neith
Another request make me test my t1 card, which has no
quality problems, but all that I get is:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793% 99.98779
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